similar to: Parking causes crash

One problem I noticed when I was testing was that if I use the flash button twice in a conversation, asterisk will crash. Does anybody have no idea why? MusicOnHold Problem: I'm really not understanding that feature. I know it works, because I can set up an Extension that plays the musiconhold music. ------------------------------------------------------------------------ ----- Somebody

When Asterisk first starts up, it will attempt to "bring up" the B channels on any PRI circuits. If you are using A@H then you can log on to the Asterisk CLI (asterisk -r) and then do "stop now" to stop asterisk. Start up Asterisk again by typing asterisk -cvvv at the Linux command line. You should see a bunch of messages on the terminal and then you'll get the Asterisk

Hi, I'm having problem with call parking. When I park call, either via transfer to xten or park digit sequence from features.conf, I hear the parking lot number read to me and the user gets transferred. However, MOH stops for the caller the moment user is transferred. The user can be retrieved by dialing the parked extension and voice resumes. If the parked user hangs up, the channel state

Hi List; I am using Firefly softphone Version 1.9.9 Build 4521 and I select IAX protocol and did the configuration in Network1 (and I checked the Active checkbox) as following: Server: 192.168.8.4 username: iax2user1 password: password In the Asterisk, I did the following configuration on the /etc/asterisk/iax.conf: [iax2user1] type=friend context=internal username=iax2user1 secret=password

hello, my questns are about few * functionality. 1) how can I make call tranfer. Not call parking. If I'm talking with some one a I want to tramnfer call to the another extension, to the other person. 2) how can I make call confernece. Not Meetme If I'm talking with some one and I want to join another person to our talk . I haven't found this in any manual :( hudecof -- mail:

I have a Pitney Bowes (USPS Postage) machine that connects via a USB modem to fill it. It connects but soon disconnects. It works fine through a standard analog phone line not connected to asterisk. I also have the 'd' option on the Dial command. exten => _1NXXNXXXXXX,1,Dial,Zap/47/BYEXTENSION||d Any ideas? John

Anyone having trouble parking calls? I haven't tried it in a while, but it seems to have stopped working. If I dial 700, I get a invalid extension. I have "include => parkedcalls" in the correct context, and I can dial 701, which tells me no call is parked there. Any ideas? Parking.conf is stock.

Hello; I am installing asterisk and dahdi on ubuntu "and I used my username bghayad to login for ubuntu and do the installation, actually I feel my problem is related to the?username and permission but I am not able how to fix it", I am facing now mainly the following two problems: The first one, asterisk is not starting automatically although I did sudo make config (for asterisk and

Hi, I have just enabled call parking on my Asterisk system and it's working well. However, I am running some Grandstream phones on my system and when you press # on them to transfer the call, the user on the other end of the line hears the tone produced by the phone, rather then Asterisk recognising that it's a command. Has anyone else has this problem and if so how did you correct it ?

Hi to all. I'm trying to use meetme on asterisk 1.4.22.1. On a debian i've compiled (as i need h323 support) openh323_v1_18_0 pwlib_v1_10_0 dahdi-linux-2.1.0.3 dahdi-tools-2.1.0.2 asterisk-1.4.22.1 All works fine, dahdi status is: asterik:/data/programmi# /etc/init.d/dahdi status ### Span 1: DAHDI_DUMMY/1 "DAHDI_DUMMY/1 (source: RTC) 1" (MASTER) asterik:/data/programmi#

I've got a situation where I need to use a transfer to the parking lot as hold, but am not going to use BLF indicators on the phone to pick up the parked calls so I need to hear the 3-digit extension after the transfer. I'm using Snom 300 phones and have tried setting a programmable button to Key Event F_TRANSFER 700, which successfully does the transfer but cuts off audio so you

I have an issue where someone will park a call, and then it will ring back to them, but because the caller-id looks like a regular inbound call, they don't know how to answer the call (these are the receptionists). I've tried to make an extention that I can transfer to that will set the caller-id, and that works, but I'm having issues. For instance.... if I do a blind transfer to an

I installed the TDM400P and installed it on a system running on Fedora Core1, please refer to the steps below. I connected an analogue phone to the FXS port. When I pick up the speaker I don't hear the dialtone although when I press a number key I hear a DTMF generated. What could be the problem? I appreciate your help 1- First I installed the TDM400P in the PCI slot and connected

I have been trying to get the dynamic parking working. For some reason when I park a call using this method the console says it is using the default parking context not the one I am trying to specidfy. It also is playing the parked extension to the caller. I am transfering the call to an extension that is doing a goto to the context below. Any ideas or examples on how to make this work.

I installed the 7/7/03 CVS today, and my customer service reps said that there were problems. So I went back to an earlier version. They could log in, but when they received a call, they hear the beep, but not the announcement or call. I am using macros, if night is on, I dial the 800# that is passed. Otherwise I play the Thank You (ty_pn), and I also pass the announcement that the agent hears

Here is a question that needs a few opinions... Recently we installed a couple of FXS gateways into a site with a SIP Proxy/Softswitch other than Asterisk. One of the things that the users on this site need to do is receive calls on single line phones on the FXS gateways and then hookflash and transfer them to other SIP users. We found that the FXS units, true to their nature as VoIP gateways,

I have one system that is crashing everytime a call is parked and I have tried recompiling the asterisk, checking out the latest SVN of 1.2 and modifying the configuration. I have identified what I think is the error and have back traces but since this is occurring on only one system I want to know what might cause this. CLI: -- SIP/xlite_brr-098d1e98 is ringing -- SIP/xlite_brr-098d1e98

Hi, I would like to use the new atxfer (native assisted transfer, see mantis item #3241) , but I've partially been able to make it work. I can receive a call and then having the caller hear MOH while talking with another extension (the one I want to transfer to), but then I can't make the caller and the trasferred talk hanging up or pressing any key combination I'm aware of. My

Hi there, Maybe this has been cared about before but I could not find any solution to this problem either in the wikis or in the list archived. If someone has found a solution before please just tell me where I can find that info..thanks :) I am using iaxComm to call other people who are either using SIP or also IAX clients, like me. All of us are connected to the same asterisk server.

Does anyone know what this means? It is in DMESG, and we have people complaining about echo. Disabled echo canceller because of tone (rx) John